Hello,
Just today found sipp and would like to use it to replay a packet
capture of a video SIP call, made with Wireshark. If it works, next, I
will try to replay captures in-parallel, with some delay, for the
purpose of testing my media server component.
Experimented with sipp for a bit. I believe that I probably
incorrectly configured playback of my capture.
I made a custom XML document. Perhaps, my exec action uses an
incorrect resource for a pcap. Something is off, as it is not working.
I must have did not edit properly the auto-generated sample XML
document. The playback of my capture, of a 90 seconds call, ends
almost immediately.
Is there a way to extract scenario from the complete packet capture or
something? Could I benefit from SIPp, just for capture playback,
without having a thorough understanding? Perhaps there is a simple
guide.
Sorry. I know little about VoIP. My goal is to do load testing, to
address a specific concern.
Thank you

I have a question about the number of concurrent calls vs peak calls. So
If I am running a scenario that says -l 30000 -r 50. Sipp is telling me
the peak is about 1900. This is the true number of actual simultaneous
calls correct? Since it doesn't seem to be theoretically possible to push
30k concurrent calls through a gig connection. I'm just trying to
understand the output better.
Thanks

Hi,
Please I am trying to create a set of scenario files to mimic behaviours of some IMS-based services(such as chat, file sharing, and geolocation exchange).
Is it possible to realise such scenarios on IMS Bench SIPp?Is it possible to run these scenrarios without using the manager on IMS Bench SIPp?
Thanks!
Segun

Dear All,
Currently I'm in situation to handle 180 with/without 100rel and 183
with/without 100rel accordingly needed to send Prack.
Basically I need to use reg expression for received 180/183 to check for
100rel and move the script execution accordingly.
Please help me in this regards.
Regards,
Shivakumar Balur

Hi Jason,
The embedded UAS script (-sn uas) does not perform registration (
http://sipp.sourceforge.net/doc/reference.html).
Even if you write your own script to perform registration and expect an
INVITE it will not be able to do both because the first request is outgoing
SIPp infers the mode as UAC.
The following should work unless you will use tcp or tls as transport:
- Implement a registration script that will only send register. reply to
authorization challenge and re-attempt.
- Then run a UAS script (perhaps the embedded one) after the
registration script is completed making sure that:
- You bind to the same port recorded as your contact during
registration.
- You use the Contact header provided during registration.
You will have to do some logging, parsing if you are behind a NAT but the
way you describe your topology you will not need it.
Regards,
-volkan
On Fri, Jun 5, 2015 at 6:57 PM, Jason Spriggs <jskswork@...> wrote:
> Hello SIPp Users,
>
> I have been running into an issue while trying to create a load tester
> using SIPp. Here is our setup:
>
> We have 2 boxes running asterisk and sipp and would like to do a load test
> of our custom tunnel plugin/module. The boxes are located on our network at
> 10.1.1.1 (Box 1) and 10.1.1.2 (Box 2). We initiate the calls from Box 2
> using the UAC built in scenario via the command:
>
> sipp 127.0.0.1 -i 10.1.1.2 -p 8832 -sn uac -l 1 -m 100 -r 1 -s 633
>
> The call is directed through our custom tunnel and ends up at Box 1. The
> dial plan then dials the SIP phone, "sipp".
>
> The issue we are running into is that the UAS server, ran via the command
> "sipp 127.0.0.1:5060 -i 10.1.1.1 -p 8823 -sn uas -au sipp -ap test -s
> sipp", will not register itself as the "sipp" sip phone on Box 1. The
> configuration files in asterisk are correct, and we have tested this by
> connecting a softphone up to the same "sipp" user, and it works.
>
> If anyone is able to help point out what in my UAS server initializer
> command is incorrect, that would be much appreciated.
>
> Many thanks,
> Jason
>
>
> ------------------------------------------------------------------------------
>
> _______________________________________________
> Sipp-users mailing list
> Sipp-users@...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>

Hello SIPp Users,
I have been running into an issue while trying to create a load tester
using SIPp. Here is our setup:
We have 2 boxes running asterisk and sipp and would like to do a load test
of our custom tunnel plugin/module. The boxes are located on our network at
10.1.1.1 (Box 1) and 10.1.1.2 (Box 2). We initiate the calls from Box 2
using the UAC built in scenario via the command:
sipp 127.0.0.1 -i 10.1.1.2 -p 8832 -sn uac -l 1 -m 100 -r 1 -s 633
The call is directed through our custom tunnel and ends up at Box 1. The
dial plan then dials the SIP phone, "sipp".
The issue we are running into is that the UAS server, ran via the command
"sipp 127.0.0.1:5060 -i 10.1.1.1 -p 8823 -sn uas -au sipp -ap test -s
sipp", will not register itself as the "sipp" sip phone on Box 1. The
configuration files in asterisk are correct, and we have tested this by
connecting a softphone up to the same "sipp" user, and it works.
If anyone is able to help point out what in my UAS server initializer
command is incorrect, that would be much appreciated.
Many thanks,
Jason

Hi Kumar,
Probably incomplete but my 2 cents:
An UA must respond to an UPDATE request immediately. In other words, there
is no time to get user input. For example, if you need to add a video
stream to an already established voice call you will probably need a
re-INVITE unless you have a special case and permissions from the
destination were already provided etc etc etc.
An early media session established in an INVITE transaction can be updated
by an UPDATE request. Clearly you can not use re-INVITE. Check out RFC6337
for offer/answer usage patterns.
UPDATE can be used without a SDP payload (target refresh purposes for
example) but not a re-INVITE. You must exhcange offer/answer even if it is
a re-play.
I hope these help.
Best,
-volkan
On Thu, Jun 4, 2015 at 8:56 AM, kumar uppu <kumar94905@...> wrote:
> HI every one,
>
>
> 1)What is the difference between Re-Invite and Update
> 2)what are the mandatory thing need to remember, while sending UPDATE and
> Re-Invite
>
> Thanks
>
> --
> Kumar
>
>
> ------------------------------------------------------------------------------
>
> _______________________________________________
> Sipp-users mailing list
> Sipp-users@...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>

Hi,
I am going to test with sipp our communication system (Avaya Session
Manager).
But I fail in setting up a call at sip message "407 Proxy Authentication
Required". Is anyone able to help me?
I tried put some messages into the xml, tried also to get it working with
sippy_cup (which helps to write xml-scenarios), but I failed again.
kind regards and thanks beforehand,
andre
--
Andre Gronwald

Hi Kumar,
Do you want to decrypt TLS pcap via wirehake? If so, you can may find the steps @ http://support.citrix.com/article/CTX116557
Thanks, v.
From: kumar uppu [mailto:kumar94905@...]
Sent: Wednesday, May 20, 2015 9:42 AM
To: sipp-users
Subject: Re: [Sipp-users] Regarding tls
HI all,
i am new to sipp,
i have tls pcap files
and i have cakey.pem file
I need to play that pcap file, but how?
Can any one please help me
Thanks
--
Kumar
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